Don't Buy What Neil Young Is Selling [GIZMODO Article]

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planet10

planet10

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johnYks said:
The CD-quality standard — which Young and HRA proponents say isn’t sufficient — wasn’t adopted randomly. It’s not a number plucked out of thin air. It’s based on sampling theory and the actual limits of human hearing. To the human ear, audio sampled above 44.1 kHz/16-bit is inaudibly different.
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The number was choosen because it was the limit of what current computer tech could do... actually a bit ahead, the 1st players were more like 14 bits.

I was taking graduate level sampling courses when i 1st saw Sony's white paper on CD. At that time i said that they would need to increase the sampling at least 4 times to get the fidelity up to where it could really start to challenge analog.

I would summarize this article as "the sky is falling, the sky is falling..."

dave
 
Max Headroom

Max Headroom

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Compact Disc Digital Audio - Wikipedia, the free encyclopedia
The selection of the sample rate was based primarily on the need to reproduce the audible frequency range of 20–20,000 Hz (20 kHz).
The Nyquist–Shannon sampling theorem states that a sampling rate of more than twice the maximum frequency of the signal to be recorded is needed, resulting in a required rate of at least 40 kHz.

The exact sampling rate of 44.1 kHz was inherited from a method of converting digital audio into an analog video signal for storage on U-matic video tape, which was the most affordable way to transfer data from the recording studio to the CD manufacturer at the time the CD specification was being developed.

The device that converts an analog audio signal into PCM audio, which in turn is changed into an analog video signal is called a PCM adaptor.
This technology could store six samples (three samples per stereo channel) in a single horizontal line.
A standard NTSC video signal has 245 usable lines per field, and 59.94 fields/s, which works out to be 44,056 samples/s/stereo channel.
Similarly, PAL has 294 lines and 50 fields, which gives 44,100 samples/s/stereo channel.

This system could store 14-bit samples with some error correction, or 16-bit samples with almost no error correction.[citation needed]
There was a long debate over the use of 14-bit (Philips) or 16-bit (Sony) quantization, and 44,056 or 44,100 samples/s (Sony) or approximately 44,000 samples/s (Philips).
When the Sony/Philips task force designed the Compact Disc, Philips had already developed a 14-bit D/A converter (DAC), but Sony insisted on 16-bit.
In the end, 16 bits and 44.1 kilosamples per second prevailed.
Philips found a way to produce 16-bit quality using its 14-bit DAC by using four times oversampling.[11]
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Dan.
 
nigelwright7557

nigelwright7557

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If broadband audio is no good why do we have amplifiers that can go well above 100KHz?
The reason is for transient response.
Also to keep up with fast changing high voltage audio signals.

I have heard lo-fi and hi-fi and the hi-fi is much better.

However we also have to take in the original recording into account, if it was done on a poor recording device then it will never get better. There are some pretty poor recordings out there, some which have been big hits but the sound is very poor.
 
D

dmills

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Joined 2008
SY said:
Really?
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Yes really, the limit is strictly greater then two samples per cycle, (Actually that the sample rate must EXCEED twice the bandwidth of the channel).

2 samples, does not work 2.01 does (Provided your reconstruction filter stopband is at Fs * 1.005/2, which is why we don't do that).

So two samples per cycle fails the test for audio, anything above that no matter now small the increase passes.

There is of course the engineering challenge of the reconstruction filter if you run overly close to the limit, which was a major problem back in the non oversampled, analogue reconstruction filter era.

All of this is well known, as is the fact that differences in mastering completely overwhelm any differences in the repro chain technology.

Now a well designed player, with a decent output stage so it can drive my cans reasonably well, loads of storage, flac support, good battery life, and a well thought out user interface would be a good thing, but I am not sure my Youngs device quite hits the spot.

Regards, Dan.
 
jan.didden

jan.didden

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nigelwright7557 said:
If broadband audio is no good why do we have amplifiers that can go well above 100KHz?
The reason is for transient response.
Also to keep up with fast changing high voltage audio signals.
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I don't think that's correct. If you have a signal limuted to 20kHz bandwidth, there can not be any transients or 'fast changing high voltage' components that contain components above 20kHz. So no need for 100kHz bw.

Jan
 
jan.didden

jan.didden

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dmills said:
Yes really, the limit is strictly greater then two samples per cycle, (Actually that the sample rate must EXCEED twice the bandwidth of the channel).

2 samples, does not work 2.01 does (Provided your reconstruction filter stopband is at Fs * 1.005/2, which is why we don't do that).

So two samples per cycle fails the test for audio, anything above that no matter now small the increase passes.

There is of course the engineering challenge of the reconstruction filter if you run overly close to the limit, which was a major problem back in the non oversampled, analogue reconstruction filter era.

All of this is well known, as is the fact that differences in mastering completely overwhelm any differences in the repro chain technology.

Now a well designed player, with a decent output stage so it can drive my cans reasonably well, loads of storage, flac support, good battery life, and a well thought out user interface would be a good thing, but I am not sure my Youngs device quite hits the spot.

Regards, Dan.
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Yes, but Okcrum stated that 4 x sampling rate would be needed and that is not correct. I agree with 2x + a bit.
And SY is well aware of that :)

jan
 
D

dmills

Member
Joined 2008
jan.didden said:
I don't think that's correct. If you have a signal limuted to 20kHz bandwidth, there can not be any transients or 'fast changing high voltage' components that contain components above 20kHz. So no need for 100kHz bw.
Jan
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True as far as it goes, but remember that bandwidth is normally specified at the -3dB point, staying flatish to 20Khz may well require the pole to be out at 100Khz, especially if you want roughly constant group delay.

My my hearing is that of a 41 year old bloke, so 20K is a distant memory, but 100K is not totally unreasonable, for all that you want to be well down by the time you hit the LW broadcast band.

Regards, Dan.
 
jan.didden

jan.didden

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dmills said:
True as far as it goes, but remember that bandwidth is normally specified at the -3dB point, staying flatish to 20Khz may well require the pole to be out at 100Khz, especially if you want roughly constant group delay.

My my hearing is that of a 41 year old bloke, so 20K is a distant memory, but 100K is not totally unreasonable, for all that you want to be well down by the time you hit the LW broadcast band.

Regards, Dan.
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Sure but lets then stay intellectually clean. Either there is no signal above 20kHz, and then there's no transients and 'fast high voltage' components either. Or there ARE transients and bla bla with spectral components above 20kHz, and then the signal bw is wider than 20kHz.

Jan
 
D

dmills

Member
Joined 2008
Even if there is nothing intentional above ~20K (Say a CD source, where such is (hopefully) known to be the case), you still need the lowpass filter at the amp input to go over far enough above 20K that the effect at 20K is minimal.

The objective after all being to limit the input bandwidth so as to keep RF out of the amplifier.

I have no problem with a power amplifier with a 100KHz or so small signal -3dB point, I probably do have a problem with a 20K small signal -3dB point.

Regards, Dan.
 
jan.didden

jan.didden

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dmills said:
Even if there is nothing intentional above ~20K (Say a CD source, where such is (hopefully) known to be the case), you still need the lowpass filter at the amp input to go over far enough above 20K that the effect at 20K is minimal.

The objective after all being to limit the input bandwidth so as to keep RF out of the amplifier.

I have no problem with a power amplifier with a 100KHz or so small signal -3dB point, I probably do have a problem with a 20K small signal -3dB point.

Regards, Dan.
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I agree with all that. My beef was with the original statement that appeared to separate signal bandwidth and the occurrence of transients and 'fast high voltage' outside of that bandwidth.

Jan
 
Charles Darwin

Charles Darwin

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Joined 2009
My understanding re amplifier bandwidth was that slew rate limiting is a source of distortion and the byproduct of a high slew rate is basically a large bandwidth.


I may well be wrong on that. I know barely enough about amps to be a danger to myself which is why tend not to screw around inside those things.
 
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